Coded data generation or conversion – Analog to or from digital conversion – Digital to analog conversion
Reexamination Certificate
2001-06-13
2002-12-03
Tokar, Michael (Department: 2819)
Coded data generation or conversion
Analog to or from digital conversion
Digital to analog conversion
C341S118000, C341S139000, C341S143000, C375S242000, C375S247000, C375S254000, C375S285000, C375S296000, C375S355000
Reexamination Certificate
active
06489909
ABSTRACT:
BACKGROUND OF THE INVENTION
The present invention relates to a method and apparatus for improving a signal-to-noise (S/N) ratio in digital-to-analog conversion of a pulse density modulated signal (PDM signal). The invention also relates to a circuit for decoding digital information in either DSD format or SACD format and providing an analog output signal having a low signal-to-noise ratio.
A variety of methods have been proposed for converting an analog signal to a digital signal and saving the digital signal on a recording medium. Also, a variety of methods have been proposed for converting a saved digital signal to an analog signal. Instead of the PCM (pulse coded modulation) scheme which is a conventional technique of digitizing an audio signal and which is used in known CD schemes, a DSD (direct stream digital) scheme has been recently proposed for analog-to-digital conversion, which is used in known SACD (super-audio CD) schemes, digital recording and digital-to-analog conversion. The main difference between the PCM technique and the DSD technique is the decimation of the delta signal modulator output in the PCM technique before recording on the CD. In the DSD technique, the audio signal is stored in the form produced by a delta-sigma ADC (analog-to-digital converter). The DSD signal is oversampled by 64Fs (Fs is the sampling frequency 44.1 kHz). On the other hand, the signal frequency of a PCM signal produced by the PCM technique and stored in a conventional CD is just Fs. Therefore, in the PCM technique, the analog-to-digital conversion of the stored PCM signal can be accomplished by various techniques, but if it is accomplished by means of a delta-sigma ADC, then decimation filtering to decimate the 64 Fs to 1 Fs is necessary.
In the DSD scheme, an analog signal is oversampled in an encoder at 2.8224 MHZ, which is 64 times higher than the sampling frequency of 44.1 kHz, by a delta-sigma type A/D converter to be converted to a 1-bit modulated digital signal. Immediately after the conversion, the 1-bit modulated digital signal is saved on a digital signal recording medium. For reproducing an analog signal from the 1-bit modulated digital signal, a sequence of the 1-bit digital signals is averaged by an analog low pass filter or the like as it is converted to an analog signal.
Generally, a signal digitized by a delta-sigma type analog-to-digital converter results in a pulse density modulated signal (PDM signal). A full-scale amplitude of a source analog signal is reduced to approximately 50% in the digitized signal. The concept of the SACD technique is using a 1-bit datastream, which has a fill-scale swing. This bit stream has just two-level information, “1”s and “0”s. The duty ratio chosen for these two levels generally is 50%/50%, and in that case, the averaged analog signal (i.e., low-pass-filtered analog signal) of full-scale cannot exceed 50% of the power supply voltage of the averaging circuitry. However, according to the definition of SACD, the full-scale of the digitized signal (i.e., the DSD signal, which is analog-to-digital converted by the oversampling modulator) is allowed to be 70% of the full-scale amplitude of a source analog signal at its maximum.
As a result, if a sequence of PDM digital signals is averaged as it is by an analog low pass filter or the like as mentioned above, a full-scale amplitude of an analog output is also reduced to 50%, resulting in a lower S/N ratio for noise in an analog circuit region of the analog low pass filter. On the other hand, a digital signal recorded in accordance with the conventional PCM scheme ensures a full-scale amplitude of 100%, so that the digital signal can be output with an amplitude approximately 100% of the amplitude of the source analog signal for an effective maximum amplitude of an output analog circuit in a digital-to-analog converter.
Thus, when an analog output circuit having equivalent noise components is used, the DSD scheme is disadvantageous over the PCM scheme and the SACD scheme because of the relatively lower S/N ratio associated with the DSD scheme.
Also, when a digital-to-analog (DA) converter is designed such that a DSD signal reproducing unit and a PCM signal reproducing unit share an analog output circuit, the two signals present unmatched full-scale amplitude levels and accordingly largely different S/N ratios.
SACD discs have a double layer of data storage, wherein a conventional PCM signal is stored in one layer as for conventional CD playback mode and the DSD signal is stored in the other layer as for SACD playback mode. It would be desirable to have a CD/SACD player which can inexpensively utilize a single playback circuit to accomplish the playback of music in either the CD mode or the SACD mode from the same SACD disc, and it would be desirable for the CD/SACD player to have a similar level of S/N ratio for output signal produced in both the CD mode and the SACD mode. One technique tries to cause the signal-to-noise ratio (SNR) of the SACD analog output (ie., playback signal) to appear to be higher than the SNR for conventional CD playback by intentionally degrading the CD SNR and enhancing the SACD SNR, in order to show that the SACD scheme is better than the conventional CD scheme. But, it would be desirable to provide a genuinely improved technique for increasing the SACD SNR.
As described above, in the DSD scheme using a conventional analog low pass filter to reproduce an analog signal, the DSD scheme is disadvantageous over the PCM scheme and SACD scheme in view of the S/N ratio.
SUMMARY OF THE INVENTION
It is therefore an object of the present invention to provide a method and apparatus for improving the S/N ratio characteristic in digital-to-analog conversion of a digital signal which has a reduced full-scale signal (as compared to the full scale value of a source analog signal), such as a pulse density modulated digital signal recorded in accordance with the DSD scheme.
It is another object of the invention to provide improved performance of a DSD digital-to-analog conversion system that is equivalent to the performance of a conventional CD system.
It is another object of the invention to provide a digital-to-analog conversion system that decodes information in either DSD format or CD (PCM) format so as to produce an equivalent analog output signal.
To achieve the above objects, the present invention provides a method of inproving an S/N ratio in digital-to-analog conversion of a pulse density modulated (PDM) digital signal represented by a waveform having a first predetermined full scale, where the PDM digital signal includes a signal component having a second predetermined fill scale, and the digital-to-analog conversion has a third predetermined full scale. The method comprises the steps of extracting the signal component from the PDM digital signal to generate an extracted signal having the second full scale; and matching the second full scale of the extracted signal with the third full scale of the digital-to-analog conversion.
According to the present invention, the step of extracting the signal component may include digital low pass filtering or decimation filtering.
Also, according to the present invention, the step of matching the full scale may include digitally multiplying the extracted signal.
The present invention also provides an apparatus for improving an S/N ratio in digital-to-analog conversion of a pulse density modulated (PDM) digital signal represented by a waveform having a first predetermined full scale, where the PDM digital signal includes a signal component having a second predetermined full scale, and the digital-to-analog conversion has a third predetermined full scale. The apparatus comprises digital filtering means for digitally filtering the PDM signal to extract the signal component, and generating a digitally filtered output signal in a digital form, where the digital filtering means has the second full scale; and full scale matching means for matching the second full scale of the digitally filtered output signal with the third full scale of th
Hamasaki Toshihiko
Nakao Shigetoshi
Brady W. James
Nguyen Khai M.
Swayze, Jr. W. Daniel
Telecky , Jr. Frederick J.
Texas Instruments Incorporated
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